[asterisk-bugs] [Asterisk 0018954]: SIP similar user/peer names are not properly identified when authentication and possibly more

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
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https://issues.asterisk.org/view.php?id=18954 
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Reported By:                rushowr
Assigned To:                
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Project:                    Asterisk
Issue ID:                   18954
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   major
Priority:                   normal
Status:                     new
Asterisk Version:           1.6.2.17 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2011-03-10 09:44 CST
Last Modified:              2011-03-17 07:49 CDT
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Summary:                    SIP similar user/peer names are not properly
identified when authentication and possibly more
Description: 
This problem has been discovered before but I can't find any reported
issues, so I'm starting a new one. Please forgive me if my Mantis searching
was flawed. 

I've experienced this with the latest 1.6.2.x and 1.6.0.x

When I have two SIP endpoints/trunks defined that have largely similar
names, Asterisk does not recognize the second definition, and therefore I
cannot make the second item place or receive a call through the Asterisk
system. 

In my example I have two Asterisk servers I'm trying to get to pass
certain calls between each other. 

The first server (STL) is configured thusly:

[MTV-peer]
username=STL-user
fromuser=STL-user
secret=EXAMPLE
; --snip--

[MTV-user]
username=STL-user
fromuser=STL-user
secret=EXAMPLE
; --snip--

[MTV2-peer]
username=STL-user
fromuser=STL-user
secret=ANOTHER
; --snip--

[MTV2-user]
username=STL-user
fromuser=STL-user
secret=ANOTHER
; --snip--

The other server (MTV2) is configured:

[STL-peer]
username=MTV2-user
fromuser=MTV2-user
secret=ANOTHER
; --snip--

[STL-user]
username=MTV2-user
fromuser=MTV2-user
secret=ANOTHER

The long and short of it is, other than MTV vs MTV2 and the passwords, MTV
and MTV2 were configured identically. MTV could send and receive calls from
STL, but not MTV2. Every call attempt would result in a FORBIDDEN response,
with no verbose output on the STL server's CLI at verbose 5, and just a
FORBIDDEN response notification on MTV2's CLI at verbose 5.

I'm attaching packet captures from "bad call attempts" between MTV2 and
STL, in an effort to help further. 

Please let me know if I can do more to assist.


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 (0132990) rushowr (reporter) - 2011-03-17 07:49
 https://issues.asterisk.org/view.php?id=18954#c132990 
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I just realized, the reason you received no log output is that there was
none. As stated in my bug report, even at level 5, there was no information
on the console or in my full log (which logs ALL levels other than DTMF
during testing in my shop). 

However, I'm putting everything back together for you so that I can
provide you with sip history and sip debug, but I believe you're going to
see the same output as the capture files, since the sip debug is just a
console output of the sip packets asterisk is working with.... more in a
few 

Issue History 
Date Modified    Username       Field                    Change               
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2011-03-17 07:49 rushowr        Note Added: 0132990                          
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